The invention relates to a method, a switching means and a telecommunication system for performing data communications between a first subscriber station of a mobile radio communication network and a second subscriber station which is connectable to said mobile radio communication network. The second subscriber station can be connected to the mobile radio communication network through the internet and/or a conventional public switched telephone network.
In particular, the invention relates to performing data communications between a mobile station of the mobile radio communication network and a second subscriber station connected to a data network (such as a data network running internet protocol (IP)). Such a second subscriber station can e.g. be an internet telephone constituted by a special internet telephone hardware or software on a workstation capable of supporting the internet protocol.
The mobile radio communication network can be a GSM-based mobile radio network (GSM: Global System of Mobile Communications), such as a D1, D2 or E-plus radio communication network in Germany.
In public switched telephone networks (PSTNs), conventionally each subscriber participating in a call has a separate telephone hand set into which the subscriber talks and from which speech is reproduced. Such conventional telephone hand sets can be radio telephones (which allow a free movement of the subscriber at home) which may use analog or digital transmission techniques even when being connected to a conventional public switched telephone network, such as German Telecom in Germany. Several conventional telephones may be interconnected in a private branch exchange system which interconnects the several telephones to one or more outside lines (conventional lines or ISDN lines).
Additionally, most subscribers nowadays also own a mobile radio telephone together with a subscription to a mobile radio communication network. The mobile radio telephone uses entirely digital transmission techniques for communicating with entities in the mobile radio communication network, e.g. with the mobile switching center thereof. Calls between mobile radio telephones or between a conventional telephone hand set and a mobile radio telephone are routed through the mobile radio communication network and the public switched telephone network.
Instead of just routing a telephone call from a public switched telephone network to a private branch exchange system (PBX), it is now also possible to route a call first to a computer network. Such a computer network can e.g. be constituted by a company intranet. Normal workstations of the computer network which have been upgraded with software or hardware to operate as a conventional telephone and/or as an internet telephone can be part of the computer network. Such computer-based telephones become more and more widespread such that it is soon anticipated that at least one party in a telephone communication is using a computer-based telephone instead of a conventional telephone hand set. Some computer networks or intranetworks as well as their interconnected workstations including the telephone software/hardware already use internet protocols for communication. This is e.g. described in an overview article by Linden decarmo xe2x80x9cInternet Telephone Standardsxe2x80x9d, PC Magazine, Feb. 18, 1997, pages 185 to 187.
Although the workstation running the telephone software and/or the mobile radio telephone of the mobile radio communication network constitute quite advanced units with respect to their digital speech coding/decoding units, the analog voicexe2x80x94when spoken into the microphone of the respective telephonexe2x80x94must still be digitized and compressed due to the restrictions imposed on the bandwidth in the available transmission channels, e.g. when the call still needs to be routed through the public switched telephone network. If e.g. a call is originated from a mobile radio telephone of a digital mobile radio communication network, the call will undergo several speech coding/decoding stages before reaching the final called subscriber station. Since each coding/decoding introduces errors, such several stages of decoding/coding processes drastically deteriorate the speech quality received at the called subscriber station.
For example, the compression protocols used in the GSM network and internet telephony are not the same and therefore there is a need for several coding/decoding (audio data compression/decompression) stages.
If internet telephones become more and more common in the future, it might be wise from the network utilization point of view to route the call from the telephone network to the IP-network (IP: Internet Protocol) from the network unit where the call is originated from. The possibility to route the call to a mobile subscriber in an IP-network directly to the serving mobile switching center opens new possibilities for cooperation between owner of IP-networks and mobile operators. For example, an international call from an internet telephone can be routed in the IP-network instead of the public switched telephone network PSTN.
Presently, several possibilities for setting up a call between a mobile radio telephone (a first subscriber station) of a mobile radio communication network and a workstation running a telephone or internet telephone software (a second subscriber station connectable to the mobile radio communication network) can be imagined, as will be explained below with reference to FIGS. 7 to 11. The invention uses all such call set-up possibilities.
a) Call from a Mobile Station to a Workstation Connected to an Internet Protocol Network
FIG. 7a shows an example of a telephone communication system where a mobile station is transmitting digitally (via a TDMA method) digitally coded speech via an antenna ANT to a switching means BSC/MSC of the mobile radio communication network PLMN. Here the switching means comprises a base station controller BSC and a mobile switching center MSC. The base station controller BSC inherently comprises an audio data (speech) coding/decoding means which is indicated with CODEC in FIG. 7a. 
Basically, as will be explained in more detail with reference to FIG. 7b, the call originated from the mobile station MS is decompressed in the audio data coding/decoding means such that a data rate of a bandwidth of 64 Kbit/s is transmitted through the public switched telephone network PSTN to the company intranet. Here, the call may be routed through a private branch exchange system PBX to the data network (IP-network), to which the individual workstations WS1, WS2, . . . , WSn are connected. Since the workstations WS1, WS2, . . . , WSn run a digital telephone network, the data arriving from the PSTN is digitally decoded after which an audio data (speech) coding/decoding means CODEC in the workstations WS perform the final digital/analog conversion. The PBX in FIG. 7a has a CODEC which can compress (i.e. code) the received speech using a protocol supported by the WS.
The call set-up in FIG. 7a is as follows. The mobile station MS sends a call set-up message including a calling number of the second subscriber station WS which is served by the PBX. Thus, the PBX must have the information that the called second subscriber stationxe2x80x94i.e. the workstation running the telephone softwarexe2x80x94wants all calls arriving for it to be diverted to the internet telephone software running on the workstation. In this case, the PBX needs protocol translation and speech coding between the PSTN and the internet telephone software running on the workstation. Since e.g. the internet telephone uses a packet-based transmission, the PBX must also perform the analog-to-packet conversion. Although the mobile station MS on one side and the workstation WS on the other side each run very advanced digital coding and transmission techniques, intermediate speech coding/decoding is still performed in the base station controller BSC and the PBX. This can deteriorate the audio data (speech) quality and slow down the transmission process as is explained in FIG. 7b. 
FIG. 7b shows the data rates during the compression/decompression for the situation in FIG. 7a. At {circle around (1)} an analog/digital converter in the mobile station converts the analog speech to a 64 kBit/s PCM signal. The CODEC of the mobile station MS compresses this PCM signal to 13 kBit/s (in case of a full rate coding) which is then transmitted to the switching means. At {circle around (3)} the CODEC of the base station controller BSC performs a decompression of the 13 kBit/s PCM data into a 64 kBit/s PCM data. At {circle around (4)} the gateway compresses the incoming 64 kBit/s PCM data for example to a 6.3 kBit/s PCM data, for example using G. 723. Finally at {circle around (5)} the workstation WS performs a decompression of the 6.3 kBit/s data, performs a D/A conversion and outputs the sound. Using two (lossy) speech codings reduces the quality of the sound. As a result the received sound in the workstation is not the same as in the GSM network. While FIG. 7b shows the situation for the GSM full rate speech coding, as its implementations are available on computers quite easily, it should be understood that the same problem likewise occurs with other speech coders specified for GSM, for example speech coders using half rate speech coding (GSM 06.20(prETS 300 581-2): xe2x80x9cEuropean Digital Cellular Telecommunication System (phase 2); half rate speech transcodingxe2x80x9d or enhanced full rate coding (GSM 06.60(prETS300 762-1)).
FIG. 8 shows another configuration of a telecommunication system, where communication between the mobile station and the workstation is carried out through the PSTN and the internet. The PSTN and the internet communicate through an internet PSTN-gateway IG. The IG is a server run by the operators of the PSTN or the PLMN. Here, the workstation WS is identified by a number which the operator has issued for this internet telephone device (i.e. for the software or hardware running on the workstation).
Thus, when setting up a call, the mobile station calls a number of the second subscriber station which the operator has issued. When the call arrives at the internet PSTN-gateway server IG, the server will set up the protocols to be used between the gateway and the internet telephone. During the call, it will do the protocol changes between the internet telephone and the normal telephone call arriving from the PSTN. Obviously, the internet and the workstation running the telephone software can communicate via packet transmission fully digitally. Nonetheless, the audio data coding/decoding (compression/decompression) is done in the base station controller BSC, before the call is routed into the PSTN.
b) Call from a Workstation/Mobile Station of the PLMN to a Workstation of the IP-network
FIG. 9 shows a telecommunication system where the mobile station MS is connected to a computer WS running an internet telephone program. The call is routed from the mobile switching center MSC through a direct access unit DAU to the internet and from there to the company intranet to which workstations WS are connected that also run internet telephone programs.
Since the PSTN is not involved in the setting up of the call, the complete call is handled as a data call, not as a speech call. The mobile radio communication network routes the data call via internet to the second subscriber station WS using the user name and IP-address (or fully qualified domain name) of the second subscriber station. The mobile radio communication network is transparent in this case and from the user""s point of view, the call is just like any other call between two computers running an internet telephone program over the internet.
However, in this case, when transferring speech data as a pure data call, only a data rate of 9.6 kBit/s can be used. Thus, a sound quality at the WS is less than in a speech call. The GSM network may offer other solutions in the near future, which will allow pure data calls with data rates up to four times of that, but their usage for speech calls will be far more expensive than normal speech calls.
c) Call from a Workstation to a Computer Connected to a Mobile Station
FIG. 10 shows a telecommunication system where a workstation WS of a data network (IP-network) incorporated in an intranet sets up a call to a computer or workstation WS connected to a mobile station MS of the mobile radio communication network PLMN. Since the internet uses a packet-orientated transmission, the PLMN has been expanded with GPRS features (GPRS: General Packet Radio Service) allowing a packet orientated transmission.
Such a GPRS-system comprises (amongst others) the serving GPRS support node SGSN and the gateway GPRS support node GGSN. Also in this case, the call is handled as a pure data call.
When a connection is to be established, the IP-number of the computer to be called (i.e. the first subscriber station WS/MS) is known by some unit in the mobile radio communication network. This unit is the gateway GPRS support node which knows to which serving GPRS support node the data should be sent. Therefore, the workstation WS first makes a connection to-this unit GGSN, which then knows how to make a connection via the mobile station MS to the computer WS connected to it.
Although such a call setup is in principle possible, it is not very cost effective nor efficient, as the GPRS network is specially designed to handle short data bursts instead of continuous long data streams like digitalized speech.
d) Call from a Workstation Connected to an IP-network via Internet/PSTN to a Mobile Station MS
FIG. 11 shows a telecommunication system where the call originated from the workstation is routed through internet and an internet PSTN-gateway server to the PLMN and thus to the mobile station MS. Here, the workstation WS contacts the gateway server IG using the IP-address of that server. Thereafter, the workstation WS provides the calling telephone number of the mobile telephone MS. The gateway IG then makes a normal PSTN-MS call using this number. During the call, the gateway IG does the decompression of the speech used in the internet telephone protocol between the workstation WS and the gateway IG and for the other direction it does the speech compression for the speech data received from the PSTN.
Thus, also in FIG. 11, two sites are present where a speech compression/decompression (encoding/decoding) takes place, namely in the BSC and the IG. This has obviously a deteriorating effect on the speech quality.
In the examples in FIGS. 7 to 11, the workstation running an internet telephone program can be any computer connected to an IP-network that is using an internet telephone program which allows the user to make calls over the IP-network to any other user or any other workstation running an internet telephone program. If the user has an access to an internet PSTN-gateway he/she can also make calls via this gateway to any other normal telephone.
The internet PSTN-gateway server IG acts as a gateway between internet and PSTN. It can establish a connection to any telephone using a normal telephone number. It can also connect a user on a workstation by knowing its IP-address and user name. This gateway may contain a database on the IP-number corresponding to a telephone number. This gateway can also be part of a private branch exchange system PBX, namely it can be part of the private branch exchange. The gateway performs a translation from uncompressed 64 kbit/s digital speech (received from the PSTN) to compressed speech using a protocol negotiated during the call set-up phase between the server IG and the internet telephone program. In the other direction, it decompresses the speech from the IP-network and forwards it to the PSTN.
Although, in FIGS. 7 to 11, the PSTN (Public Switched Telephone Network) is used for routing the call to the correct destination on the basis of a telephone number, in a call set-up phase, it is possible to inquire the capabilities of the exchanges along the route (e.g. Do all the exchanges provide support for a certain service etc.). It should further be noted that even in a configuration like FIG. 11, all traffic is communicated digitally, although there may still be some analog exchanges present.
As was explained with reference to FIGS. 7 to 11 above, it is envisaged that calls can be set up between mobile stations and workstations running a telephone software through various paths, e.g. directly through the internet or indirectly through the PSTN and then through the internet. It is also possible to connect a computer running a telephone software to the mobile station and then likewise set up a call to a workstation running an internet telephone software.
However, due to several compressions/decompressions, the speech quality deteriorates on the transmission path between the mobile station and the workstation.
Thus, the object of the present invention is
the provision of a method, a switching means and a telecommunication system, which maintain a high speech quality between a first subscriber station of the mobile radio communication network and a second subscriber station connectable to the mobile radio communication network, in particular for cases when the second subscriber station uses a telephone software running on a computer.
This object is solved by a method for performing data communications between a first subscriber station (MS) of a mobile radio communication network (PLMN) and a second subscriber station (WS) connectable to said mobile radio communication network (PLMN), wherein at least said first subscriber station (MS) comprises an audio data encoding/decoding means (CODEC), comprising the following steps: sending a call set-up message from said first or second subscriber station (MS) to a switching means (BSC, MSC/VLR) of said mobile radio communication network (PLMN) to set up a call between said first and second subscriber stations (WS, MS); determining on the basis of said call setup message whether said second subscriber station (WS) is of a type also comprising an audio data encoding/decoding means (CODEC); setting up a call between said first and second subscriber station (WS); switching off an audio data encoding/decoding means (DECOD) in said switching means (BSC, MSC/VLR), if said second subscriber station (WS) also comprises an audio data encoding/decoding means (CODEC); and encoding/decoding audio data at said first and second subscriber station (MS, WS) using said respective encoding/decoding means (CODEC) and communicating said coded audio data through said switching means (BSC, MSC/VLR) without applying an audio data coding/decoding thereto in said switching means (BSC, MSC/VLR).
Furthermore, this object is solved by a switching means (BSC, MSC/VLR) of a mobile radio communication network (PLMN) for communicating data between a first subscriber station (MS) of said mobile radio communication network (PLMN) and a second subscriber station (WS) connectable to said mobile radio communication network (PLMN), wherein at least said first subscriber station (MS) comprises an audio data encoding/decoding means (CODEC), comprising: an audio data encoding/decoding means (CODEC) including first state in which digital audio data received from said first/second subscriber station is encoded/decoded; and a second state in which digital audio data received from said first/second subscriber station is passed without applying an audio encoding/decoding thereto; subscriber station type determining means (SSTDM) for determining whether said second subscriber station (WS) is of a type comprising an audio data encoding/decoding means (CODEC) on the basis of a call-set-up message sent from said first or second subscriber station (MS) in a call set-up phase; and control means (CNTRL) for switching said audio data encoding/decoding means (CODEC) of said switching means into said second state when said subscriber station type determining means (SSTDM) determines that said second subscriber station (WS) is of type also having an independent audio data encoding/decoding means (DECOD).
Furthermore, this object is solved by a telecommunication system (PLMN, PSTN, INTRANET; INTERNET, IP-NET), comprising: a mobile radio communication network (PLMN) to which at least one first subscriber station (MS) having an audio data coding/decoding means (CODEC) is connected; an intranet to which at least one second subscriber station (WS) is connected; and internet and/or a public switched telephone network (PSTN) connected between said mobile radio communication network (PLMN) and said intranet; wherein said mobile radio communication network (PLMN) comprises a switching means (BSC, MSC, VLR) for communicating data between one first and one second subscriber station (MS, WS) including: an audio data encoding/decoding means (CODEC) including a first state in which digital audio data received from said first/second subscriber station is encoded/decoded; and a second state in which digital audio data received from said first/second subscriber station is passed without applying an audio data encoding/decoding thereto; subscriber station type determining means (SSTDM) for determining whether said second subscriber station (WS) is of a type comprising an audio data encoding/decoding means (CODEC) on the basis of a call-set-up message sent from said first or second subscriber station (MS) in a call set-up phase; and control means (CNTRL) for switching said audio data encoding/decoding means (CODEC) of said switching means into said second state when said subscriber station type determining means (SSTDM) determines that said second subscriber station (WS) is of type also having an independent audio data encoding/decoding means (CODEC).
Therefore, according to the invention, the switching means of the mobile radio communication network comprises an audio data encoding/decoding means which has two states. In a first state, the digital audio data coming from the mobile station or being transmitted to the mobile station undergoes normal compression/decompression. In a second state, the digital audio data from/to the mobile station passes through the switching means without applying any coding/decoding to it.
Whether the first or second state of the audio data encoding/decoding means is selected is determined by a subscriber station type determining means. Whenever the subscriber station type determining means determines that the second subscriber station is a workstation computer running a telephone software being connected to an Internet protocol network, the subscriber station type determining means recognizes that digital uncompressed data can be sent/transmitted directly to the workstation on the basis of this determination, the control means switches the audio data encoding/decoding means of the switching means into its second state, such that the data between the first and second subscriber station (the mobile station and the workstation) is freely communicated without applying any additional and unnecessary speech encoding/decoding to it. Since unnecessary speech compression/decompression is avoided, the speech quality is improved.
Depending on the configuration of the telecommunication network, the switching means of the mobile radio communication network either receives a message from a calling workstation that the workstation does not need compressed/decompressed speech data or the switching means itself can enquire whether the called second subscriber station is one that supports a telephone software on a computer. Preferably, the switching means of the mobile radio communication network, the gateway exchange of the public switched telephone network or the internet PSTN-gateway are connected to a respective memory where the specific details of the first and second subscriber stations are recorded. Therefore, independently as to whether the mobile station or the workstation originates the call, it can always be ensured that there is no unnecessary speech compression/decompression after the call has been set up.
Further solutions of the above object are listed in claims 33-46. Further advantageous embodiments and improvements of the invention are listed in the dependent claims. Hereinafter, the invention will be described with reference to its embodiments and the attached drawings. In the drawings, the same or similar reference numerals designate the same or similar parts throughout.